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Unified Diff: webrtc/media/engine/webrtcvoiceengine_unittest.cc

Issue 1813763005: Updated structures and functions for setting the max bitrate limit to take rtc::Optional<int> Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Code review feedback Created 4 years, 9 months ago
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Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 30be07d21398f97572f453ed08bb930c805f5986..a4fdcf242cefd3afe2971725821e6e98a75e6553 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -189,12 +189,12 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
// |expected_result| is the expected result from SetMaxSendBandwidth().
// |expected_bitrate| is the expected audio bitrate afterward.
void TestSendBandwidth(const cricket::AudioCodec& codec,
- int max_bitrate,
+ rtc::Optional<int> max_bitrate,
bool expected_result,
int expected_bitrate) {
cricket::AudioSendParameters parameters;
parameters.codecs.push_back(codec);
- parameters.max_bandwidth_bps = max_bitrate;
+ parameters.max_bitrate_bps = max_bitrate;
EXPECT_EQ(expected_result, channel_->SetSendParameters(parameters));
int channel_num = voe_.GetLastChannel();
@@ -693,13 +693,13 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
// bitrate is <= 0.
// ISAC, default bitrate == 32000.
- TestSendBandwidth(kIsacCodec, 0, true, 32000);
+ TestSendBandwidth(kIsacCodec, rtc::Optional<int>(), true, 32000);
// PCMU, default bitrate == 64000.
- TestSendBandwidth(kPcmuCodec, -1, true, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(), true, 64000);
// opus, default bitrate == 64000.
- TestSendBandwidth(kOpusCodec, -1, true, 64000);
+ TestSendBandwidth(kOpusCodec, rtc::Optional<int>(), true, 64000);
}
TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
@@ -708,12 +708,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
// Test that the bitrate of a multi-rate codec is always the maximum.
// ISAC, default bitrate == 32000.
- TestSendBandwidth(kIsacCodec, 128000, true, 128000);
- TestSendBandwidth(kIsacCodec, 16000, true, 16000);
+ TestSendBandwidth(kIsacCodec, rtc::Optional<int>(128000), true, 128000);
+ TestSendBandwidth(kIsacCodec, rtc::Optional<int>(16000), true, 16000);
// opus, default bitrate == 64000.
- TestSendBandwidth(kOpusCodec, 96000, true, 96000);
- TestSendBandwidth(kOpusCodec, 48000, true, 48000);
+ TestSendBandwidth(kOpusCodec, rtc::Optional<int>(96000), true, 96000);
+ TestSendBandwidth(kOpusCodec, rtc::Optional<int>(48000), true, 48000);
}
TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
@@ -723,13 +723,13 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
// if it's bigger than the fixed rate.
// PCMU, fixed bitrate == 64000.
- TestSendBandwidth(kPcmuCodec, 0, true, 64000);
- TestSendBandwidth(kPcmuCodec, 1, false, 64000);
- TestSendBandwidth(kPcmuCodec, 128000, true, 64000);
- TestSendBandwidth(kPcmuCodec, 32000, false, 64000);
- TestSendBandwidth(kPcmuCodec, 64000, true, 64000);
- TestSendBandwidth(kPcmuCodec, 63999, false, 64000);
- TestSendBandwidth(kPcmuCodec, 64001, true, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(), true, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(1), false, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(128000), true, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(32000), false, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(64000), true, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(63999), false, 64000);
+ TestSendBandwidth(kPcmuCodec, rtc::Optional<int>(64001), true, 64000);
}
TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
@@ -737,7 +737,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
const int kDesiredBitrate = 128000;
cricket::AudioSendParameters parameters;
parameters.codecs = engine_.codecs();
- parameters.max_bandwidth_bps = kDesiredBitrate;
+ parameters.max_bitrate_bps = rtc::Optional<int>(kDesiredBitrate);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->AddSendStream(
@@ -762,12 +762,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) {
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec));
EXPECT_EQ(64000, codec.rate);
- send_parameters_.max_bandwidth_bps = 128000;
+ send_parameters_.max_bitrate_bps = rtc::Optional<int>(128000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec));
EXPECT_EQ(64000, codec.rate);
- send_parameters_.max_bandwidth_bps = 128;
+ send_parameters_.max_bitrate_bps = rtc::Optional<int>(128);
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, codec));
EXPECT_EQ(64000, codec.rate);
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